[alsa-devel] [PATCH 1/5] [RFC] ALSA ASoC driver for TLVaic23b audio codec
Arun KS
arunks at mistralsolutions.com
Mon Sep 29 11:45:04 CEST 2008
ASOC Audio driver for TLVaic23b audio codec
Signed-off-by: Arun KS <arunks at mistralsolutions.com>
---
sound/soc/codecs/Kconfig | 4 +
sound/soc/codecs/Makefile | 2 +
sound/soc/codecs/tlv320aic23.c | 634 ++++++++++++++++++++++++++++++++++++++++
sound/soc/codecs/tlv320aic23.h | 143 +++++++++
4 files changed, 783 insertions(+), 0 deletions(-)
create mode 100644 sound/soc/codecs/tlv320aic23.c
create mode 100644 sound/soc/codecs/tlv320aic23.h
diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig
index b1a5eed..7be0b37 100644
--- a/sound/soc/codecs/Kconfig
+++ b/sound/soc/codecs/Kconfig
@@ -55,3 +55,7 @@ config SND_SOC_TWL4030
tristate
depends on SND_SOC && TWL4030_CORE
+config SND_SOC_TLV320AIC23
+ tristate
+ depends on I2C
+
diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile
index a519ced..adb5aa6 100644
--- a/sound/soc/codecs/Makefile
+++ b/sound/soc/codecs/Makefile
@@ -11,6 +11,7 @@ snd-soc-wm9713-objs := wm9713.o
snd-soc-cs4270-objs := cs4270.o
snd-soc-tlv320aic3x-objs := tlv320aic3x.o
snd-soc-twl4030-objs := twl4030.o
+snd-soc-tlv320aic23-objs := tlv320aic23.o
obj-$(CONFIG_SND_SOC_AC97_CODEC) += snd-soc-ac97.o
obj-$(CONFIG_SND_SOC_AK4535) += snd-soc-ak4535.o
@@ -25,3 +26,4 @@ obj-$(CONFIG_SND_SOC_WM9713) += snd-soc-wm9713.o
obj-$(CONFIG_SND_SOC_CS4270) += snd-soc-cs4270.o
obj-$(CONFIG_SND_SOC_TLV320AIC3X) += snd-soc-tlv320aic3x.o
obj-$(CONFIG_SND_SOC_TWL4030) += snd-soc-twl4030.o
+obj-$(CONFIG_SND_SOC_TLV320AIC23) += snd-soc-tlv320aic23.o
diff --git a/sound/soc/codecs/tlv320aic23.c b/sound/soc/codecs/tlv320aic23.c
new file mode 100644
index 0000000..eefeb2d
--- /dev/null
+++ b/sound/soc/codecs/tlv320aic23.c
@@ -0,0 +1,634 @@
+/*
+ * ALSA SoC TLV320AIC23 codec driver
+ *
+ * Author: Arun KS, <arunks at mistralsolutions.com>
+ * Copyright: (C) 2008 Mistral Solutions Pvt Ltd.,
+ *
+ * Based on sound/soc/codecs/tlv320aic23.c by Vladimir Barinov
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ *
+ * Notes:
+ * The AIC23 is a driver for a low power stereo audio
+ * codec aic23
+ *
+ * The machine layer should disable unsupported inputs/outputs by
+ * snd_soc_dapm_disable_pin(codec, "LHPOUT"), etc.
+ */
+
+#include <linux/module.h>
+#include <linux/moduleparam.h>
+#include <linux/init.h>
+#include <linux/delay.h>
+#include <linux/pm.h>
+#include <linux/i2c.h>
+#include <linux/platform_device.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+#include <sound/initval.h>
+
+#include "tlv320aic23.h"
+
+#define AUDIO_NAME "aic23"
+#define AIC23_VERSION "0.1"
+
+/* codec private data */
+struct aic23_priv {
+ unsigned int sysclk;
+ int master;
+};
+
+/*
+ * AIC23 register cache
+ */
+static const u16 aic23_reg[AIC23_CACHEREGNUM] = {
+ 0x0097, 0x0097, 0x00F9, 0x00F9, /* 0 */
+ 0x001A, 0x0004, 0x0007, 0x0001, /* 4 */
+ 0x0020, 0x0000, 0x0000, 0x0000, /* 8 */
+ 0x0000, 0x0000, 0x0000, 0x0000, /* 12 */
+};
+
+/*
+ * read aic23 register cache
+ */
+static inline unsigned int aic23_read_reg_cache(struct snd_soc_codec *codec,
+ unsigned int reg)
+{
+ u16 *cache = codec->reg_cache;
+ if (reg >= AIC23_CACHEREGNUM)
+ return -1;
+ return cache[reg];
+}
+
+/*
+ * write aic23 register cache
+ */
+static inline void aic23_write_reg_cache(struct snd_soc_codec *codec,
+ u8 reg, u16 value)
+{
+ u16 *cache = codec->reg_cache;
+ if (reg >= AIC23_CACHEREGNUM)
+ return;
+ cache[reg] = value;
+}
+
+/*
+ * write to the aic23 register space
+ */
+static int aic23_write(struct snd_soc_codec *codec, unsigned int reg,
+ unsigned int value)
+{
+
+ u8 data;
+
+ /* TLV320AIC23 has 7 bit address and 9 bits of data
+ * so we need to switch one data bit into reg and rest
+ * of data into val
+ */
+
+ if ((reg < 0 || reg > 9) && (reg != 15)) {
+ printk(KERN_WARNING "Invalid register R%d\n", reg);
+ return -1;
+ }
+
+ data = (reg << 1) | (value >> 8 & 0x01);
+
+ aic23_write_reg_cache(codec, reg, value);
+
+ if (codec->hw_write(codec->control_data, (u8) data,
+ (u8) (value & 0xff)) == 0)
+ return 0;
+
+ printk(KERN_ERR "I2C: cannot write %03x to register R%d\n", value, reg);
+
+ return -EIO;
+}
+
+static const struct snd_kcontrol_new aic23_snd_controls[] = {
+ SOC_DOUBLE_R("PCM Playback Volume", LEFT_CHANNEL_VOLUME_ADDR,
+ RIGHT_CHANNEL_VOLUME_ADDR, 0, 0x7f, 0),
+ SOC_SINGLE("PCM Playback Switch", DIGITAL_AUDIO_CONTROL_ADDR, 3, 0x01,
+ 1),
+ SOC_DOUBLE_R("Line Input Mute", LEFT_LINE_VOLUME_ADDR,
+ RIGHT_LINE_VOLUME_ADDR, 7, 0x01, 0),
+ SOC_DOUBLE_R("Line Input Volume", LEFT_LINE_VOLUME_ADDR,
+ RIGHT_LINE_VOLUME_ADDR, 0, 0x1f, 0),
+ SOC_SINGLE("Mic Mute", ANALOG_AUDIO_CONTROL_ADDR, 1, 0x01, 0),
+ SOC_SINGLE("Mic Booster Switch", ANALOG_AUDIO_CONTROL_ADDR, 0, 0x01, 0),
+
+};
+
+/* add non dapm controls */
+static int aic23_add_controls(struct snd_soc_codec *codec)
+{
+
+ int err, i;
+
+ for (i = 0; i < ARRAY_SIZE(aic23_snd_controls); i++) {
+ err = snd_ctl_add(codec->card,
+ snd_soc_cnew(&aic23_snd_controls[i],
+ codec, NULL));
+ if (err < 0)
+ return err;
+ }
+
+ return 0;
+
+}
+static const char *aic23_rec_src[] = { "Line", "Mic" };
+
+static const struct soc_enum aic23_enum[] = {
+ SOC_ENUM_SINGLE(ANALOG_AUDIO_CONTROL_ADDR, 2, 2, aic23_rec_src),
+};
+
+static const struct snd_kcontrol_new aic23_rec_src_mux_controls =
+SOC_DAPM_ENUM("Route", aic23_enum[0]);
+
+/* PGA Mixer controls for Line and Mic switch */
+static const struct snd_kcontrol_new aic23_pga_mixer_controls[] = {
+ SOC_DAPM_SINGLE("Line Switch", POWER_DOWN_CONTROL_ADDR, 0, 1, 1),
+ SOC_DAPM_SINGLE("Mic Switch", POWER_DOWN_CONTROL_ADDR, 1, 1, 1),
+};
+
+static const struct snd_soc_dapm_widget aic23_dapm_widgets[] = {
+ SND_SOC_DAPM_DAC("DAC", "Playback", POWER_DOWN_CONTROL_ADDR, 3, 1),
+ SND_SOC_DAPM_PGA("OUT", POWER_DOWN_CONTROL_ADDR, 4, 1, NULL, 0),
+
+ SND_SOC_DAPM_ADC("ADC", "Capture", POWER_DOWN_CONTROL_ADDR, 2, 1),
+ SND_SOC_DAPM_MIXER("PGA Mixer", SND_SOC_NOPM, 0, 0,
+ &aic23_pga_mixer_controls[0],
+ ARRAY_SIZE(aic23_pga_mixer_controls)),
+ SND_SOC_DAPM_MUX("Capture Source", SND_SOC_NOPM, 0, 0,
+ &aic23_rec_src_mux_controls),
+
+ SND_SOC_DAPM_OUTPUT("LHPOUT"),
+ SND_SOC_DAPM_OUTPUT("RHPOUT"),
+ SND_SOC_DAPM_OUTPUT("LOUT"),
+ SND_SOC_DAPM_OUTPUT("ROUT"),
+ SND_SOC_DAPM_INPUT("LLINEIN"),
+ SND_SOC_DAPM_INPUT("RLINEIN"),
+ SND_SOC_DAPM_INPUT("MICIN"),
+};
+
+static const struct snd_soc_dapm_route intercon[] = {
+ {"OUT", NULL, "DAC"},
+
+ {"LHPOUT", NULL, "OUT"},
+ {"RHPOUT", NULL, "OUT"},
+
+ {"LOUT", NULL, "OUT"},
+ {"ROUT", NULL, "OUT"},
+
+ {"Capture Source", "Line", "LLINEIN"},
+ {"Capture Source", "Line", "RLINEIN"},
+
+ {"Capture Source", "Mic", "MICIN"},
+
+ {"PGA Mixer", "Line Switch", "Capture Source"},
+ {"PGA Mixer", "Mic Switch", "Capture Source"},
+
+ {"ADC", NULL, "PGA Mixer"},
+
+};
+
+/* aic23 related */
+static const struct aic23_samplerate_reg_info
+ rate_reg_info[NUMBER_SAMPLE_RATES_SUPPORTED] = {
+ {4000, 0x06, 1}, /* 4000 */
+ {8000, 0x06, 0}, /* 8000 */
+ {16000, 0x0C, 1}, /* 16000 */
+ {22050, 0x11, 1}, /* 22050 */
+ {24000, 0x00, 1}, /* 24000 */
+ {32000, 0x0C, 0}, /* 32000 */
+ {44100, 0x11, 0}, /* 44100 */
+ {48000, 0x00, 0}, /* 48000 */
+ {88200, 0x1F, 0}, /* 88200 */
+ {96000, 0x0E, 0}, /* 96000 */
+};
+
+static int aic23_add_widgets(struct snd_soc_codec *codec)
+{
+ snd_soc_dapm_new_controls(codec, aic23_dapm_widgets,
+ ARRAY_SIZE(aic23_dapm_widgets));
+
+ /* set up audio path interconnects */
+ snd_soc_dapm_add_routes(codec, intercon, ARRAY_SIZE(intercon));
+
+ snd_soc_dapm_new_widgets(codec);
+ return 0;
+}
+
+static int aic23_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_device *socdev = rtd->socdev;
+ struct snd_soc_codec *codec = socdev->codec;
+ u16 iface_reg, data;
+ u8 count = 0;
+
+ iface_reg =
+ aic23_read_reg_cache(codec,
+ DIGITAL_AUDIO_FORMAT_ADDR) & ~(0x03 << 2);
+
+ /* Search for the right sample rate */
+ /* Verify what happens if the rate is not supported
+ * now it goes to 96Khz */
+ while ((rate_reg_info[count].sample_rate != params_rate(params)) &&
+ (count < ARRAY_SIZE(rate_reg_info))) {
+ count++;
+ }
+
+ data = (rate_reg_info[count].divider << CLKIN_SHIFT) |
+ (rate_reg_info[count].control << BOSR_SHIFT) | USB_CLK_ON;
+
+ aic23_write(codec, SAMPLE_RATE_CONTROL_ADDR, data);
+
+ switch (params_format(params)) {
+ case SNDRV_PCM_FORMAT_S16_LE:
+ break;
+ case SNDRV_PCM_FORMAT_S20_3LE:
+ iface_reg |= (0x01 << 2);
+ break;
+ case SNDRV_PCM_FORMAT_S24_LE:
+ iface_reg |= (0x02 << 2);
+ break;
+ case SNDRV_PCM_FORMAT_S32_LE:
+ iface_reg |= (0x03 << 2);
+ break;
+ }
+ aic23_write(codec, DIGITAL_AUDIO_FORMAT_ADDR, iface_reg);
+
+ return 0;
+}
+
+static int aic23_mute(struct snd_soc_dai *dai, int mute)
+{
+ struct snd_soc_codec *codec = dai->codec;
+ u16 reg;
+
+ reg =
+ aic23_read_reg_cache(codec,
+ DIGITAL_AUDIO_CONTROL_ADDR) & ~DACM_MUTE;
+
+ if (mute)
+ aic23_write(codec, DIGITAL_AUDIO_CONTROL_ADDR, reg | DACM_MUTE);
+
+ else
+ aic23_write(codec, DIGITAL_AUDIO_CONTROL_ADDR, reg);
+
+ return 0;
+}
+
+static int aic23_set_dai_sysclk(struct snd_soc_dai *codec_dai,
+ int clk_id, unsigned int freq, int dir)
+{
+ struct snd_soc_codec *codec = codec_dai->codec;
+ struct aic23_priv *aic23 = codec->private_data;
+
+ aic23->sysclk = freq;
+ return 0;
+}
+
+static int aic23_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt)
+{
+ struct snd_soc_codec *codec = codec_dai->codec;
+ struct aic23_priv *aic23 = codec->private_data;
+ u16 iface_reg;
+
+ iface_reg =
+ aic23_read_reg_cache(codec, DIGITAL_AUDIO_FORMAT_ADDR) & (~0x03);
+
+ /* set master/slave audio interface */
+ switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
+ case SND_SOC_DAIFMT_CBM_CFM:
+ aic23->master = 1;
+ iface_reg |= MS_MASTER;
+ break;
+ case SND_SOC_DAIFMT_CBS_CFS:
+ aic23->master = 0;
+ break;
+ default:
+ return -EINVAL;
+
+ }
+
+ /* interface format */
+ switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+ case SND_SOC_DAIFMT_I2S:
+ iface_reg |= FOR_I2S;
+ break;
+ case SND_SOC_DAIFMT_DSP_A:
+ iface_reg |= FOR_DSP;
+ break;
+ case SND_SOC_DAIFMT_RIGHT_J:
+ break;
+ case SND_SOC_DAIFMT_LEFT_J:
+ iface_reg |= FOR_LJUST;
+ break;
+ default:
+ return -EINVAL;
+
+ }
+
+ aic23_write(codec, DIGITAL_AUDIO_FORMAT_ADDR, iface_reg);
+
+ return 0;
+}
+
+static int aic23_set_bias_level(struct snd_soc_codec *codec,
+ enum snd_soc_bias_level level)
+{
+ u16 reg;
+
+ switch (level) {
+ case SND_SOC_BIAS_ON:
+ reg = aic23_read_reg_cache(codec, POWER_DOWN_CONTROL_ADDR);
+ aic23_write(codec, POWER_DOWN_CONTROL_ADDR,
+ reg & (~DEVICE_POWER_OFF));
+ /* Activate interface */
+ reg = aic23_read_reg_cache(codec, DIGITAL_INTERFACE_ACT_ADDR);
+ aic23_write(codec, DIGITAL_INTERFACE_ACT_ADDR, reg | ACT_ON);
+ break;
+ case SND_SOC_BIAS_PREPARE:
+ break;
+ case SND_SOC_BIAS_STANDBY:
+ reg = aic23_read_reg_cache(codec, POWER_DOWN_CONTROL_ADDR);
+ aic23_write(codec, POWER_DOWN_CONTROL_ADDR,
+ reg | DEVICE_POWER_OFF);
+ /* Deactivate interface */
+ reg = aic23_read_reg_cache(codec, DIGITAL_INTERFACE_ACT_ADDR);
+ aic23_write(codec, DIGITAL_INTERFACE_ACT_ADDR, reg & (~ACT_ON));
+
+ break;
+ case SND_SOC_BIAS_OFF:
+ break;
+ }
+
+ codec->bias_level = level;
+
+ return 0;
+}
+
+#define AIC23_RATES SNDRV_PCM_RATE_8000_96000
+#define AIC23_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \
+ SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_S32_LE)
+
+struct snd_soc_dai aic23_dai = {
+ .name = "aic23",
+ .playback = {
+ .stream_name = "Playback",
+ .channels_min = 2,
+ .channels_max = 2,
+ .rates = AIC23_RATES,
+ .formats = AIC23_FORMATS,},
+ .capture = {
+ .stream_name = "Capture",
+ .channels_min = 2,
+ .channels_max = 2,
+ .rates = AIC23_RATES,
+ .formats = AIC23_FORMATS,},
+ .ops = {
+ .hw_params = aic23_hw_params,
+ },
+ .dai_ops = {
+ .digital_mute = aic23_mute,
+ .set_sysclk = aic23_set_dai_sysclk,
+ .set_fmt = aic23_set_dai_fmt,
+ }
+};
+EXPORT_SYMBOL_GPL(aic23_dai);
+
+static int aic23_suspend(struct platform_device *pdev, pm_message_t state)
+{
+ struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+ struct snd_soc_codec *codec = socdev->codec;
+
+ aic23_set_bias_level(codec, SND_SOC_BIAS_OFF);
+
+ return 0;
+}
+
+static int aic23_resume(struct platform_device *pdev)
+{
+ struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+ struct snd_soc_codec *codec = socdev->codec;
+ int i;
+ u8 data[2];
+ u8 *cache = codec->reg_cache;
+
+ /* Sync reg_cache with the hardware */
+ for (i = 0; i < ARRAY_SIZE(aic23_reg); i++) {
+ data[0] = i;
+ data[1] = cache[i];
+ codec->hw_write(codec->control_data, data, 2);
+ }
+
+ aic23_set_bias_level(codec, codec->suspend_bias_level);
+
+ return 0;
+}
+
+/*
+ * initialise the AIC23 driver
+ * register the mixer and dsp interfaces with the kernel
+ */
+static int aic23_init(struct snd_soc_device *socdev)
+{
+ struct snd_soc_codec *codec = socdev->codec;
+ int ret = 0;
+ u16 reg;
+
+ codec->name = "aic23";
+ codec->owner = THIS_MODULE;
+ codec->read = aic23_read_reg_cache;
+ codec->write = aic23_write;
+ codec->set_bias_level = aic23_set_bias_level;
+ codec->dai = &aic23_dai;
+ codec->num_dai = 1;
+ codec->reg_cache_size = ARRAY_SIZE(aic23_reg);
+ codec->reg_cache = kmemdup(aic23_reg, sizeof(aic23_reg), GFP_KERNEL);
+ if (codec->reg_cache == NULL)
+ return -ENOMEM;
+
+ /* Reset codec */
+ aic23_write(codec, RESET_CONTROL_ADDR, 0);
+
+ /* register pcms */
+ ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1);
+ if (ret < 0) {
+ printk(KERN_ERR "aic23: failed to create pcms\n");
+ goto pcm_err;
+ }
+
+ aic23_write(codec, DIGITAL_AUDIO_CONTROL_ADDR, DEEMP_44K);
+
+ /* Unmute input */
+ reg = aic23_read_reg_cache(codec, LEFT_LINE_VOLUME_ADDR);
+ aic23_write(codec, LEFT_LINE_VOLUME_ADDR,
+ (reg & (~LIM_MUTED)) | (LRS_ENABLED));
+ reg = aic23_read_reg_cache(codec, RIGHT_LINE_VOLUME_ADDR);
+ aic23_write(codec, RIGHT_LINE_VOLUME_ADDR,
+ (reg & (~LIM_MUTED)) | LRS_ENABLED);
+ reg = aic23_read_reg_cache(codec, ANALOG_AUDIO_CONTROL_ADDR);
+ aic23_write(codec, ANALOG_AUDIO_CONTROL_ADDR,
+ (reg | DAC_SELECTED) & (~MICM_MUTED) & (~BYPASS_ON));
+
+ /* Default output volume */
+ aic23_write(codec, LEFT_CHANNEL_VOLUME_ADDR,
+ DEFAULT_OUTPUT_VOLUME & OUTPUT_VOLUME_MASK);
+ aic23_write(codec, RIGHT_CHANNEL_VOLUME_ADDR,
+ DEFAULT_OUTPUT_VOLUME & OUTPUT_VOLUME_MASK);
+
+ /* off, with power on */
+ aic23_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
+
+ aic23_add_controls(codec);
+ aic23_add_widgets(codec);
+ ret = snd_soc_register_card(socdev);
+ if (ret < 0) {
+ printk(KERN_ERR "aic23: failed to register card\n");
+ goto card_err;
+ }
+
+ return ret;
+
+card_err:
+ snd_soc_free_pcms(socdev);
+ snd_soc_dapm_free(socdev);
+pcm_err:
+ kfree(codec->reg_cache);
+ return ret;
+}
+static struct snd_soc_device *aic23_socdev;
+
+#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
+/*
+ * If the i2c layer weren't so broken, we could pass this kind of data
+ * around
+ */
+static int aic23_codec_probe(struct i2c_client *i2c,
+ const struct i2c_device_id *i2c_id)
+{
+ struct snd_soc_device *socdev = aic23_socdev;
+ struct snd_soc_codec *codec = socdev->codec;
+ int ret;
+
+ if (!i2c_check_functionality(i2c->adapter, I2C_FUNC_SMBUS_BYTE_DATA))
+ return -EINVAL;
+
+
+ i2c_set_clientdata(i2c, codec);
+ codec->control_data = i2c;
+
+ ret = aic23_init(socdev);
+ if (ret < 0) {
+ printk(KERN_ERR "aic23: failed to initialise AIC23\n");
+ goto err;
+ }
+ return ret;
+
+err:
+ kfree(codec);
+ kfree(i2c);
+ return ret;
+}
+static int __exit aic23_i2c_remove(struct i2c_client *i2c)
+{
+
+ put_device(&i2c->dev);
+ return 0;
+}
+
+static const struct i2c_device_id tlvaic23b_id[] = {
+ {"tlvaic23b", 0},
+ {}
+};
+
+MODULE_DEVICE_TABLE(i2c, tlvaic23b_id);
+
+static struct i2c_driver aic23b_i2c_driver = {
+ .driver = {
+ .name = "tlvaic23b",
+ },
+ .probe = aic23_codec_probe,
+ .remove = __exit_p(aic23_i2c_remove),
+ .id_table = tlvaic23b_id,
+};
+
+#endif
+
+static int aic23_probe(struct platform_device *pdev)
+{
+ struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+ struct snd_soc_codec *codec;
+ struct aic23_priv *aic23;
+ int ret = 0;
+
+ printk(KERN_INFO "AIC23 Audio Codec %s\n", AIC23_VERSION);
+
+ codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL);
+ if (codec == NULL)
+ return -ENOMEM;
+
+ aic23 = kzalloc(sizeof(struct aic23_priv), GFP_KERNEL);
+ if (aic23 == NULL) {
+ kfree(codec);
+ return -ENOMEM;
+ }
+
+ codec->private_data = aic23;
+ socdev->codec = codec;
+ mutex_init(&codec->mutex);
+ INIT_LIST_HEAD(&codec->dapm_widgets);
+ INIT_LIST_HEAD(&codec->dapm_paths);
+
+ aic23_socdev = socdev;
+#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
+ codec->hw_write = (hw_write_t) i2c_smbus_write_byte_data;
+ codec->hw_read = NULL;
+ ret = i2c_add_driver(&aic23b_i2c_driver);
+ if (ret != 0)
+ printk(KERN_ERR "can't add i2c driver");
+#else
+ /* Add other interfaces here */
+#endif
+ return ret;
+}
+
+static int aic23_remove(struct platform_device *pdev)
+{
+ struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+ struct snd_soc_codec *codec = socdev->codec;
+
+ /* power down chip */
+ if (codec->control_data)
+ aic23_set_bias_level(codec, SND_SOC_BIAS_OFF);
+
+ snd_soc_free_pcms(socdev);
+ snd_soc_dapm_free(socdev);
+#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
+ i2c_del_driver(&aic23b_i2c_driver);
+#endif
+ kfree(codec->private_data);
+ kfree(codec->reg_cache);
+ kfree(codec);
+
+ return 0;
+}
+struct snd_soc_codec_device soc_codec_dev_aic23 = {
+ .probe = aic23_probe,
+ .remove = aic23_remove,
+ .suspend = aic23_suspend,
+ .resume = aic23_resume,
+};
+EXPORT_SYMBOL_GPL(soc_codec_dev_aic23);
+
+MODULE_DESCRIPTION("ASoC TLV320AIC23 codec driver");
+MODULE_AUTHOR("Arun KS");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/tlv320aic23.h b/sound/soc/codecs/tlv320aic23.h
new file mode 100644
index 0000000..9c3808c
--- /dev/null
+++ b/sound/soc/codecs/tlv320aic23.h
@@ -0,0 +1,143 @@
+/*
+ * ALSA SoC TLV320AIC23 codec driver
+ *
+ * Author: Arun KS, <arunks at mistralsolutions.com>
+ * Copyright: (C) 2008 Mistral Solutions Pvt Ltd
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ */
+
+#ifndef _AIC23_H
+#define _AIC23_H
+
+#define AIC23_CACHEREGNUM 16
+/* Codec TLV320AIC23 */
+#define LEFT_LINE_VOLUME_ADDR 0x00
+#define RIGHT_LINE_VOLUME_ADDR 0x01
+#define LEFT_CHANNEL_VOLUME_ADDR 0x02
+#define RIGHT_CHANNEL_VOLUME_ADDR 0x03
+#define ANALOG_AUDIO_CONTROL_ADDR 0x04
+#define DIGITAL_AUDIO_CONTROL_ADDR 0x05
+#define POWER_DOWN_CONTROL_ADDR 0x06
+#define DIGITAL_AUDIO_FORMAT_ADDR 0x07
+#define SAMPLE_RATE_CONTROL_ADDR 0x08
+#define DIGITAL_INTERFACE_ACT_ADDR 0x09
+#define RESET_CONTROL_ADDR 0x0F
+
+/* Left (right) line input volume control register */
+#define LRS_ENABLED 0x0100
+#define LIM_MUTED 0x0080
+#define LIV_DEFAULT 0x0017
+#define LIV_MAX 0x001f
+#define LIV_MIN 0x0000
+
+/* Left (right) channel headphone volume control register */
+#define LZC_ON 0x0080
+#define LHV_DEFAULT 0x0079
+#define LHV_MAX 0x007f
+#define LHV_MIN 0x0000
+
+/* Analog audio path control register */
+#define STA_REG(x) ((x)<<6)
+#define STE_ENABLED 0x0020
+#define DAC_SELECTED 0x0010
+#define BYPASS_ON 0x0008
+#define INSEL_MIC 0x0004
+#define MICM_MUTED 0x0002
+#define MICB_20DB 0x0001
+
+/* Digital audio path control register */
+#define DACM_MUTE 0x0008
+#define DEEMP_32K 0x0002
+#define DEEMP_44K 0x0004
+#define DEEMP_48K 0x0006
+#define ADCHP_ON 0x0001
+
+/* Power control down register */
+#define DEVICE_POWER_OFF 0x0080
+#define CLK_OFF 0x0040
+#define OSC_OFF 0x0020
+#define OUT_OFF 0x0010
+#define DAC_OFF 0x0008
+#define ADC_OFF 0x0004
+#define MIC_OFF 0x0002
+#define LINE_OFF 0x0001
+
+/* Digital audio interface register */
+#define MS_MASTER 0x0040
+#define LRSWAP_ON 0x0020
+#define LRP_ON 0x0010
+#define IWL_16 0x0000
+#define IWL_20 0x0004
+#define IWL_24 0x0008
+#define IWL_32 0x000C
+#define FOR_I2S 0x0002
+#define FOR_DSP 0x0003
+#define FOR_LJUST 0x0001
+
+/* Sample rate control register */
+#define CLKOUT_HALF 0x0080
+#define CLKIN_HALF 0x0040
+#define BOSR_384fs 0x0002 /* BOSR_272fs when in USB mode */
+#define USB_CLK_ON 0x0001
+#define SR_MASK 0xf
+#define CLKOUT_SHIFT 7
+#define CLKIN_SHIFT 6
+#define SR_SHIFT 2
+#define BOSR_SHIFT 1
+
+/* Digital interface register */
+#define ACT_ON 0x0001
+
+/* Define to set the AIC23 as the master w.r.t McBSP */
+#define AIC23_MASTER
+
+#define NUMBER_SAMPLE_RATES_SUPPORTED 10
+
+/*
+ * AUDIO related MACROS
+ */
+#ifndef DEFAULT_BITPERSAMPLE
+#define DEFAULT_BITPERSAMPLE 16
+#endif
+
+#define DEFAULT_SAMPLE_RATE 44100
+#define CODEC_CLOCK 12000000
+#define AUDIO_MCBSP OMAP_MCBSP1
+
+#define DEFAULT_OUTPUT_VOLUME 0x70
+#define DEFAULT_INPUT_VOLUME 0x10 /* 0 ==> mute line in */
+
+#define OUTPUT_VOLUME_MIN LHV_MIN
+#define OUTPUT_VOLUME_MAX LHV_MAX
+#define OUTPUT_VOLUME_RANGE (OUTPUT_VOLUME_MAX - OUTPUT_VOLUME_MIN)
+#define OUTPUT_VOLUME_MASK OUTPUT_VOLUME_MAX
+
+#define INPUT_VOLUME_MIN LIV_MIN
+#define INPUT_VOLUME_MAX LIV_MAX
+#define INPUT_VOLUME_RANGE (INPUT_VOLUME_MAX - INPUT_VOLUME_MIN)
+#define INPUT_VOLUME_MASK INPUT_VOLUME_MAX
+
+#define SIDETONE_MASK 0x1c0
+#define SIDETONE_0 0x100
+#define SIDETONE_6 0x000
+#define SIDETONE_9 0x040
+#define SIDETONE_12 0x080
+#define SIDETONE_18 0x0c0
+
+#define DEFAULT_ANALOG_AUDIO_CONTROL (DAC_SELECTED | STE_ENABLED | \
+ INSEL_MIC | MICB_20DB)
+
+struct aic23_samplerate_reg_info {
+ u32 sample_rate;
+ u8 control; /* SR3, SR2, SR1, SR0 and BOSR */
+ u8 divider /* if 0 CLKIN = MCLK, if 1 CLKIN = MCLK/2 */
+};
+
+extern struct snd_soc_dai aic23_dai;
+extern struct snd_soc_codec_device soc_codec_dev_aic23;
+
+#endif /* _AIC23_H */
+
--
1.5.3.4
--
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